Antonio Sanz
2003-Nov-25 16:24 UTC
[Asterisk-Users] About sound modules in Asterisk. And call gnophone-asterisk-h323
Hi,
First at alll, I beg your pardon because maybe I explained bad my questions
(because my low level english)
I have asterisk 0.5.0, asterisk-oh323-0.5.6, openh323-1.12.2 and pwlib 1.5.2
compiled and installed.
I have modules alsa 0.9.8 compiled and installed
My PC has an audio card ac97 chipset intel i810 in its motherboard.
I want to use asterisk in this way:
|PC linux | --ethernet-- |asterisk| --internet-- |gatekeeper| -- |PSTN|
|
|PC win| -----ethernet--- |
A PC's network connected to * via ethernet with gnophone clients or
iaxcomm.ASterisk is connected to a gatekeeper and the gatekeeper sends the
calls to the PSTN.
In order to do that, maybe I don't need soundcard in *. Coul you please
confirm it to me?
In the other hand. in astersk configuration examples there are references
to calls that finish in the PBX and can be used as voice mail or to give
anouncements or simply to make an operator hang-off. I understand that a
soundcard is needed for some of these options
When I ran asterisk loading alsa module. * gave me an error, and finished.
Now, I ran it with module oss and it starts, but it gives me this warning
[chan_oss.so] => (OSS Console Channel Driver)
== Console is full duplex
== Registered channel type 'Console' (OSS Console Channel Driver)
== Parsing '/etc/asterisk/oss.conf': Found
[res_adsi.so]WARNING[3076]: File chan_oss.c, Line 232 (sound_thread): Read
error on sound device: Resource temporarily unavailable
I don't know how to to do that soundcard i8x0 works with OSS, I only know
to make it work with ALSA, but for the tasks I want to do maybe I don't
need soundcard, for that reason I would like to comment another problem I
have in the configuration I am testing
In the configuration file 0h323 I have defined a gatekeeper, and when *
starts, it is well registered in that gatekeeper.
When I make a call since gnophone (Linux) or iaxcomm (win98) to a PSTN
telephone number, the call goes well, the PSTN telephone rings, but when I
hang-off nothing is heard in any address.
The * call logis:
CLI> -- Accepting AUTHENTICATED call from 10.16.96.149, requested format
= 2, actual format = 2
-- Executing Dial("IAX[marko@10.16.96.149:5036]/1",
"OH323/BYEXTENSION@80.98.105.178") in new stack
-- Called 913014647@80.98.105.178
-- H323:25026 answered IAX[marko@10.16.96.149:5036]/1
-- Hungup 'H323:25026'
== Spawn extension (default, 913014647, 1) exited non-zero on
'IAX[marko@10.16.96.149:5036]/1'
-- Hungup 'IAX[marko@10.16.96.149:5036]/1'
Un the file extensions.conf I have an input for all te calls to go through
the gatekeeper:
exten => _91XXXXXXX,1,Dial,OH323/BYEXTENSION@80.98.105.178
I don't know if it is mandatory to define here the gatekeeper address
(although I have already done in the oh323.conf), but in this same
distribution list I read this configuration example (with this one, the
call works, but nothing is heard):
When the call is done, I see packets between gnophone and asterisk
bidirectionally: 0.16.96.148 (asterisk) 10.16.96.149(gnophone)
namor:/home/sanz# tcpdump -n -i eth0 udp and host 10.16.96.148
tcpdump: listening on eth0
Estabishment phase:
17:36:55.650506 10.16.96.149.5036 > 10.16.96.148.5036: udp 104 (DF)
[tos0x10]
17:36:55.654766 10.16.96.149.5036 > 10.16.96.148.5036: udp 12 (DF) [tos
0x10]
17:36:55.674871 10.16.96.148.5036 > 10.16.96.149.5036: udp 12 (DF) [tos
0x10]
17:36:55.675328 10.16.96.148.5036 > 10.16.96.149.5036: udp 69 (DF) [tos
0x10]
17:36:55.675504 10.16.96.148.5036 > 10.16.96.149.5036: udp 12 (DF) [tos
0x10]
17:36:55.676650 10.16.96.149.5036 > 10.16.96.148.5036: udp 12 (DF) [tos
0x10]
17:36:55.690271 10.16.96.148.5036 > 10.16.96.149.5036: udp 12 (DF) [tos
0x10]
17:36:55.691258 10.16.96.149.5036 > 10.16.96.148.5036: udp 12 (DF) [tos
0x10]
17:37:00.314595 10.16.96.149.5036 > 10.16.96.148.5036: udp 55 (DF) [tos
0x10]
17:37:00.314758 10.16.96.148.5036 > 10.16.96.149.5036: udp 12 (DF) [tos
0x10]
17:37:00.315792 10.16.96.148.5036 > 10.16.96.149.5036: udp 23 (DF) [tos
0x10]
17:37:00.316550 10.16.96.149.5036 > 10.16.96.148.5036: udp 12 (DF) [tos
0x10]
17:37:00.881438 10.16.96.148.5036 > 10.16.96.149.5036: udp 12 (DF) [tos
0x10]
17:37:00.882394 10.16.96.149.5036 > 10.16.96.148.5036: udp 12 (DF) [tos
0x10]
17:37:00.887593 10.16.96.149.5036 > 10.16.96.148.5036: udp 45 (DF) [tos
0x10]
17:37:00.887812 10.16.96.148.5036 > 10.16.96.149.5036: udp 12 (DF) [tos
0x10]
17:37:00.890902 10.16.96.148.5036 > 10.16.96.149.5036: udp 45 (DF) [tos
0x10]
17:37:00.891565 10.16.96.149.5036 > 10.16.96.148.5036: udp 12 (DF) [tos
0x10]
voice
17:37:00.904894 10.16.96.149.5036 > 10.16.96.148.5036: udp 37 (DF) [tos
0x10]
17:37:00.910874 10.16.96.148.5036 > 10.16.96.149.5036: udp 37 (DF) [tos
0x10]
17:37:00.922539 10.16.96.149.5036 > 10.16.96.148.5036: udp 37 (DF) [tos
0x10]
17:37:00.931519 10.16.96.148.5036 > 10.16.96.149.5036: udp 37 (DF) [tos
0x10]
17:37:00.940179 10.16.96.149.5036 > 10.16.96.148.5036: udp 37 (DF) [tos
0x10]
17:37:00.951466 10.16.96.148.5036 > 10.16.96.149.5036: udp 37 (DF) [tos
0x10]
17:37:00.957994 10.16.96.149.5036 > 10.16.96.148.5036: udp 37 (DF) [tos
0x10]
.............................................................................
But un the gatekeeper communication process I only see packets in one address_
namor:/home/sanz# tcpdumpatm -n -i atm0
tcpdumpatm: listening on atm0
17:56:13.659867 80.98.102.206.10004 > 80.98.105.178.1720: S
1165965604:1165965604(0) win 18280 <mss 9140,sackOK,timestamp
4428183360,nop,wscale 0> (DF)
17:56:13.660000 80.98.105.178.1720 > 80.98.102.206.10004:
S4211354764:4211354764(0) ack 1165965605 win 8760 <mss 1460> (DF)
17:56:13.660914 80.98.102.206.10004 > 80.98.105.178.1720: . ack 1 win 18280
(DF)
17:56:13.687009 80.98.102.206.10004 > 80.98.105.178.1720: P 1:363(362) ack
1 win 18280 (DF)
17:56:13.820000 80.98.105.178.1720 > 80.98.102.206.10004: . ack 363 win
8398 (DF)
17:56:14.090000 80.98.105.178.1720 > 80.98.102.206.10004: P 1:158(157)
ack363 win 8398 (DF)
17:56:14.097194 80.98.102.206.10004 > 80.98.105.178.1720: . ack 158 win
19296 (DF)
voice
17:56:14.090000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172
17:56:14.110000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172
17:56:14.130000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172
17:56:14.150000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172
17:56:14.170000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172
17:56:14.190000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172
17:56:14.210000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172
17:56:14.230000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172
17:56:14.250000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172
17:56:14.270000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172
17:56:14.290000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172
Only voice packets Gategkeeper ---> Asterisk. No packets Asterisk -->
Gatekeeper
Best regards
Michael Manousos
2003-Nov-26 02:31 UTC
[Asterisk-Users] About sound modules in Asterisk. And call gnophone-asterisk-h323
What is your H.323 configuration? You must provide the contents of oh323.conf and the relevant lines of extensions.conf (preferably off-list). Michael. Antonio Sanz wrote:> Hi, > > First at alll, I beg your pardon because maybe I explained bad my > questions (because my low level english) > > I have asterisk 0.5.0, asterisk-oh323-0.5.6, openh323-1.12.2 and pwlib > 1.5.2 > > compiled and installed. > > I have modules alsa 0.9.8 compiled and installed > > My PC has an audio card ac97 chipset intel i810 in its motherboard. > > I want to use asterisk in this way: > > |PC linux | --ethernet-- |asterisk| --internet-- |gatekeeper| -- > |PSTN| > | > |PC win| -----ethernet--- | > > A PC's network connected to * via ethernet with gnophone clients or > iaxcomm.ASterisk is connected to a gatekeeper and the gatekeeper sends > the calls to the PSTN. > > In order to do that, maybe I don't need soundcard in *. Coul you please > confirm it to me? > > In the other hand. in astersk configuration examples there are > references to calls that finish in the PBX and can be used as voice mail > or to give anouncements or simply to make an operator hang-off. I > understand that a soundcard is needed for some of these options > > When I ran asterisk loading alsa module. * gave me an error, and > finished. Now, I ran it with module oss and it starts, but it gives me > this warning > > [chan_oss.so] => (OSS Console Channel Driver) > == Console is full duplex > == Registered channel type 'Console' (OSS Console Channel Driver) > == Parsing '/etc/asterisk/oss.conf': Found > [res_adsi.so]WARNING[3076]: File chan_oss.c, Line 232 (sound_thread): Read > error on sound device: Resource temporarily unavailable > > I don't know how to to do that soundcard i8x0 works with OSS, I only > know to make it work with ALSA, but for the tasks I want to do maybe I > don't need soundcard, for that reason I would like to comment another > problem I have in the configuration I am testing > > In the configuration file 0h323 I have defined a gatekeeper, and when * > starts, it is well registered in that gatekeeper. > > When I make a call since gnophone (Linux) or iaxcomm (win98) to a PSTN > telephone number, the call goes well, the PSTN telephone rings, but when > I hang-off nothing is heard in any address. > > The * call logis: > > CLI> -- Accepting AUTHENTICATED call from 10.16.96.149, requested > format > = 2, actual format = 2 > -- Executing Dial("IAX[marko@10.16.96.149:5036]/1", > "OH323/BYEXTENSION@80.98.105.178") in new stack > -- Called 913014647@80.98.105.178 > -- H323:25026 answered IAX[marko@10.16.96.149:5036]/1 > -- Hungup 'H323:25026' > == Spawn extension (default, 913014647, 1) exited non-zero on > 'IAX[marko@10.16.96.149:5036]/1' > -- Hungup 'IAX[marko@10.16.96.149:5036]/1' > > > Un the file extensions.conf I have an input for all te calls to go > through the gatekeeper: > > exten => _91XXXXXXX,1,Dial,OH323/BYEXTENSION@80.98.105.178 > > I don't know if it is mandatory to define here the gatekeeper address > (although I have already done in the oh323.conf), but in this same > distribution list I read this configuration example (with this one, the > call works, but nothing is heard): > > When the call is done, I see packets between gnophone and asterisk > bidirectionally: 0.16.96.148 (asterisk) 10.16.96.149(gnophone) > > namor:/home/sanz# tcpdump -n -i eth0 udp and host 10.16.96.148 > tcpdump: listening on eth0 > > Estabishment phase: > > 17:36:55.650506 10.16.96.149.5036 > 10.16.96.148.5036: udp 104 (DF) > [tos0x10] > 17:36:55.654766 10.16.96.149.5036 > 10.16.96.148.5036: udp 12 (DF) [tos > 0x10] > 17:36:55.674871 10.16.96.148.5036 > 10.16.96.149.5036: udp 12 (DF) [tos > 0x10] > 17:36:55.675328 10.16.96.148.5036 > 10.16.96.149.5036: udp 69 (DF) [tos > 0x10] > 17:36:55.675504 10.16.96.148.5036 > 10.16.96.149.5036: udp 12 (DF) [tos > 0x10] > 17:36:55.676650 10.16.96.149.5036 > 10.16.96.148.5036: udp 12 (DF) [tos > 0x10] > 17:36:55.690271 10.16.96.148.5036 > 10.16.96.149.5036: udp 12 (DF) [tos > 0x10] > 17:36:55.691258 10.16.96.149.5036 > 10.16.96.148.5036: udp 12 (DF) [tos > 0x10] > 17:37:00.314595 10.16.96.149.5036 > 10.16.96.148.5036: udp 55 (DF) [tos > 0x10] > 17:37:00.314758 10.16.96.148.5036 > 10.16.96.149.5036: udp 12 (DF) [tos > 0x10] > 17:37:00.315792 10.16.96.148.5036 > 10.16.96.149.5036: udp 23 (DF) [tos > 0x10] > 17:37:00.316550 10.16.96.149.5036 > 10.16.96.148.5036: udp 12 (DF) [tos > 0x10] > 17:37:00.881438 10.16.96.148.5036 > 10.16.96.149.5036: udp 12 (DF) [tos > 0x10] > 17:37:00.882394 10.16.96.149.5036 > 10.16.96.148.5036: udp 12 (DF) [tos > 0x10] > 17:37:00.887593 10.16.96.149.5036 > 10.16.96.148.5036: udp 45 (DF) [tos > 0x10] > 17:37:00.887812 10.16.96.148.5036 > 10.16.96.149.5036: udp 12 (DF) [tos > 0x10] > 17:37:00.890902 10.16.96.148.5036 > 10.16.96.149.5036: udp 45 (DF) [tos > 0x10] > 17:37:00.891565 10.16.96.149.5036 > 10.16.96.148.5036: udp 12 (DF) [tos > 0x10] > > voice > > 17:37:00.904894 10.16.96.149.5036 > 10.16.96.148.5036: udp 37 (DF) [tos > 0x10] > 17:37:00.910874 10.16.96.148.5036 > 10.16.96.149.5036: udp 37 (DF) [tos > 0x10] > 17:37:00.922539 10.16.96.149.5036 > 10.16.96.148.5036: udp 37 (DF) [tos > 0x10] > 17:37:00.931519 10.16.96.148.5036 > 10.16.96.149.5036: udp 37 (DF) [tos > 0x10] > 17:37:00.940179 10.16.96.149.5036 > 10.16.96.148.5036: udp 37 (DF) [tos > 0x10] > 17:37:00.951466 10.16.96.148.5036 > 10.16.96.149.5036: udp 37 (DF) [tos > 0x10] > 17:37:00.957994 10.16.96.149.5036 > 10.16.96.148.5036: udp 37 (DF) [tos > 0x10] > ............................................................................. > > > But un the gatekeeper communication process I only see packets in one > address_ > > namor:/home/sanz# tcpdumpatm -n -i atm0 > tcpdumpatm: listening on atm0 > 17:56:13.659867 80.98.102.206.10004 > 80.98.105.178.1720: S > 1165965604:1165965604(0) win 18280 <mss 9140,sackOK,timestamp > 4428183360,nop,wscale 0> (DF) > 17:56:13.660000 80.98.105.178.1720 > 80.98.102.206.10004: > S4211354764:4211354764(0) ack 1165965605 win 8760 <mss 1460> (DF) > 17:56:13.660914 80.98.102.206.10004 > 80.98.105.178.1720: . ack 1 win > 18280 (DF) > 17:56:13.687009 80.98.102.206.10004 > 80.98.105.178.1720: P 1:363(362) > ack 1 win 18280 (DF) > 17:56:13.820000 80.98.105.178.1720 > 80.98.102.206.10004: . ack 363 win > 8398 (DF) > 17:56:14.090000 80.98.105.178.1720 > 80.98.102.206.10004: P 1:158(157) > ack363 win 8398 (DF) > 17:56:14.097194 80.98.102.206.10004 > 80.98.105.178.1720: . ack 158 win > 19296 (DF) > > voice > > 17:56:14.090000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172 > 17:56:14.110000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172 > 17:56:14.130000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172 > 17:56:14.150000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172 > 17:56:14.170000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172 > 17:56:14.190000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172 > 17:56:14.210000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172 > 17:56:14.230000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172 > 17:56:14.250000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172 > 17:56:14.270000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172 > 17:56:14.290000 80.98.105.179.45084 > 80.98.102.206.10010: udp 172 > > > Only voice packets Gategkeeper ---> Asterisk. No packets Asterisk > --> Gatekeeper > > Best regards > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users