Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in the * ). But now problem comes in the outbound as when i use a extension like exten=>12,1,Dial(OH323/12) Then the call goes through but i don't hear any voice. So my two problems are 1.Why asterisk gives seg. fault when i dial exten on 711 codec from ATA 2.Why can't i hear voice from * to ATA when i use 723 in ATA. for 2nd i think that there is mismatch between the codecs so can we change the priority order of the codecs used in the * or Oh323 and if yes, then how? Please ask if any further Input is required. Rgds Manoj K Gupta ----- Original Message ----- From: "Michael Manousos" <manousos@inaccessnetworks.com> To: <asterisk-users@lists.digium.com> Sent: Thursday, August 07, 2003 4:54 PM Subject: Re: [Asterisk-Users] Warning Messages> > Inband DTMF detection works only on G.711 frames. > It seems that the codec negotiation phase ended up > with the GSM codec, so you get these messages. > A quick fix would be to comment out this informative > line in 'dsp.c' file. > > > Michael. > > > surajee@infotechs.lk wrote: > > hi, > > > > i have connected a SNOM 200 to the asterisk. hereare my settings,> > > > Codecs > > ------- > > > > Default codec - g.711u/g.711a > > Packet size - 20ms > > Negotiation - Interoperable > > Type - 160 > > > > DTMF > > ---- > > > > Inband - negotiate > > Outband - negotiate > > Payload Type - 101 > > > > when a call comes to the SNOM or when making anoutdial, following warning> > messages are coming on asteisk, > > > > WARNING[1209214400]: File dsp.c, Line 1198(ast_dsp_process): Unable to detect process 2 frames> > WARNING[1209214400]: File dsp.c, Line 1198(ast_dsp_process): Unable to detect process 2 frames> > WARNING[1209214400]: File dsp.c, Line 1198(ast_dsp_process): Unable to detect process 2 frames> > ... > > WARNING[1209214400]: File dsp.c, Line 1198(ast_dsp_process): Unable to detect process 2 frames> > WARNING[1209214400]: File dsp.c, Line 1198(ast_dsp_process): Unable to detect process 2 frames> > > > these warning messages come a lot, but still u canhave a normal voice> > conversation. > > > > but this warning messages are very irritative.. > > > > does anybody has an idea on this? > > > > Thanks inadvance, > > Surajee > > > > --------------This mail sent throughOmniBIS.com--------------> > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users> > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com
Michael Manousos
2003-Aug-08 03:26 UTC
[Asterisk-Users] Problem -ATA-711-723-Oh323-Asterisk
Sip Rtp wrote:> Hi List, > > I am facing the reverse problem as stated here.I am > using ATA 186 to make > and recieve call to * through OH323 driver. > When I use G711 codec in the ATA to make call then > then as soon as i dial an > extension the * crashes with 'segmentation fault'.More information is needed. You should provide a backtrace of the core file, the screen log of Asterisk (generated when executed with "asterisk -vvvcdg"), your oh323.conf and the important sections of extensions.conf.> But the same scenerio works fine when i use 723 codec > in the ATA .I can dial > the number and extension very well/(I have 723 support > in the * ). > But now problem comes in the outbound as when i use a > extension like > exten=>12,1,Dial(OH323/12) > Then the call goes through but i don't hear any voice. > So my two problems are > 1.Why asterisk gives seg. fault when i dial exten on > 711 codec from ATA > 2.Why can't i hear voice from * to ATA when i use 723 > in ATA. > for 2nd i think that there is mismatch between the > codecs so can we change > the priority order of the codecs used in the * or > Oh323 and if yes, then > how? > > Please ask if any further Input is required. > > Rgds > Manoj K Gupta >Michael.
Hello Michael,
Here is the information which you asked for.
Please look into it..If you need more info tell.
I am using the following call scenerio..
I am dialing to PBX from openphone by dialing a PSTN
number connected to *
through development kit of digium.
then i press 12 as the extension to dial for ATA
connected to GNUGK.
Thanks for the time
Rgds
Sip Rtp
Written by Mark Spencer <markster@linux-support.net>
========================================================================DEBUG[1074447520]:
File config.c, Line 712
(__ast_load): Parsing
/etc/asterisk/logger.conf
Asterisk Event Logger Started
/var/log/asterisk/event_log
== Manager registered action Ping
== Manager registered action Logoff
== Manager registered action Hangup
== Manager registered action Status
== Manager registered action Redirect
== Manager registered action Originate
== Manager registered action MailboxStatus
== Manager registered action Command
== Manager registered action ExtensionState
== Manager registered action AbsoluteTimeout
== Manager registered action MailboxCount
Asterisk Management interface listening on port 5038
== RTP Allocating from port range 10000 -> 20000
Asterisk PBX Core Initializing
Registering builtin applications:
[AbsoluteTimeout]
== Registered application 'AbsoluteTimeout'
[Answer]
== Registered application 'Answer'
[BackGround]
== Registered application 'BackGround'
[Busy]
== Registered application 'Busy'
[Congestion]
== Registered application 'Congestion'
[DigitTimeout]
== Registered application 'DigitTimeout'
[Goto]
== Registered application 'Goto'
[GotoIf]
== Registered application 'GotoIf'
[GotoIfTime]
== Registered application 'GotoIfTime'
[Hangup]
== Registered application 'Hangup'
[NoOp]
== Registered application 'NoOp'
[Prefix]
== Registered application 'Prefix'
[ResponseTimeout]
== Registered application 'ResponseTimeout'
[Ringing]
== Registered application 'Ringing'
[SayNumber]
== Registered application 'SayNumber'
[SayDigits]
== Registered application 'SayDigits'
[SetAccount]
== Registered application 'SetAccount'
[SetGlobalVar]
== Registered application 'SetGlobalVar'
[SetLanguage]
== Registered application 'SetLanguage'
[SetVar]
== Registered application 'SetVar'
[StripMSD]
== Registered application 'StripMSD'
[Suffix]
== Registered application 'Suffix'
[Wait]
== Registered application 'Wait'
Asterisk Dynamic Loader Starting:
[chan_modem.so] => (Generic Voice Modem Driver)
== Loading modem driver chan_modem_aopen.so =>
(A/Open (Rockwell Chipset)
ITU-2 VoiceModem Driver)
[res_musiconhold.so] => (Music On Hold Resource)
== Registered application 'MusicOnHold'
== Registered application 'WaitMusicOnHold'
== Registered application 'SetMusicOnHold'
[res_adsi.so] => (Call Parking Resource)
[res_parking.so] => (Call Parking Resource)
== Registered application 'ParkedCall'
[res_crypto.so] => (Cryptographic Digital Signatures)
-- Loaded PUBLIC key 'iaxtel'
[res_indications.so] => (Indications Configuration)
-- Registered indication country 'us'
-- Registered indication country 'au'
-- Registered indication country 'fr'
-- Registered indication country 'de'
-- Registered indication country 'nl'
-- Registered indication country 'uk'
-- Setting default indication country to 'us'
== Registered application 'Playtones'
== Registered application 'StopPlaytones'
[res_monitor.so] => (Call Monitoring Resource)
== Registered application 'Monitor'
== Registered application 'StopMonitor'
== Registered application 'ChangeMonitor'
== Manager registered action Monitor
== Manager registered action StopMonitor
== Manager registered action ChangeMonitor
[chan_iax.so] => (Inter Asterisk eXchange)
== Manager registered action IAXpeers
== Using TOS bits 16
== IAX Ready and Listening on 0.0.0.0 port 5036
[chan_sip.so] => (Session Initiation Protocol (SIP))
== SIP Listening on 0.0.0.0:5060
== Using TOS bits 0
[skipping chan_oss.so]
[chan_modem_bestdata.so] => (BestData (Conexant V.90
Chipset) VoiceModem
Driver)
[chan_modem_i4l.so] => (ISDN4Linux Emulated Modem
Driver)
[chan_agent.so] => (Agent Proxy Channel)
== Registered application 'AgentLogin'
== Registered application 'AgentCallbackLogin'
[chan_mgcp.so] => (Media Gateway Control Protocol
(MGCP))
== MGCP Listening on 0.0.0.0:2427
[chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
== Manager registered action IAXpeers
WARNING[1074447520]: File chan_iax2.c, Line 5061
(set_config): Ignoring port
for now
== Using TOS bits 16
== IAX Ready and Listening on 0.0.0.0 port 4569
[chan_local.so] => (Local Proxy Channel)
[chan_phone.so] => (Linux Telephony API Support)
[chan_zap.so] => (Zapata Telephony w/PRI)
-- Registered channel 1, FXS Kewlstart signalling
-- Registered channel 2, FXS Kewlstart signalling
-- Registered channel 3, FXS Kewlstart signalling
-- Registered channel 4, FXS Kewlstart signalling
-- Registered channel 5, FXS Kewlstart signalling
-- Registered channel 6, FXS Kewlstart signalling
-- Registered channel 7, FXS Kewlstart signalling
-- Registered channel 8, FXS Kewlstart signalling
WARNING[1074447520]: File chan_zap.c, Line 6654
(load_module): Ignoring
rxwink
WARNING[1074447520]: File chan_zap.c, Line 6654
(load_module): Ignoring
cancelforward
WARNING[1074447520]: File chan_zap.c, Line 6654
(load_module): Ignoring
echocanelwhenbridged
-- Registered channel 9, FXO Kewlstart signalling
-- Registered channel 10, FXO Kewlstart signalling
-- Registered channel 11, FXO Kewlstart signalling
-- Registered channel 12, FXO Kewlstart signalling
-- Registered channel 13, FXO Kewlstart signalling
-- Registered channel 14, FXO Kewlstart signalling
-- Registered channel 15, FXO Kewlstart signalling
-- Registered channel 16, FXO Kewlstart signalling
-- Registered channel 17, FXO Kewlstart signalling
-- Registered channel 18, FXO Kewlstart signalling
-- Registered channel 19, FXO Kewlstart signalling
-- Registered channel 20, FXO Kewlstart signalling
-- Registered channel 21, FXO Kewlstart signalling
-- Registered channel 22, FXO Kewlstart signalling
-- Registered channel 23, FXO Kewlstart signalling
-- Registered channel 24, FXO Kewlstart signalling
[pbx_config.so] => (Text Extension Configuration)
-- Setting global variable 'CONSOLE' to
'Zap/1/125844'
-- Setting global variable 'HEMANT' to 'OH323/007'
-- Setting global variable 'GOPESH' to 'OH323/008'
-- Setting global variable 'MANISH' to 'OH323/009'
-- Setting global variable 'IAXINFO' to 'guest'
-- Setting global variable 'TRUNK' to 'Zap/g3'
-- Including context 'longdistance' in context
'international'
-- Including context 'trunkint' in context
'international'
-- Including context 'local' in context
'longdistance'
-- Including context 'trunkld' in context
'longdistance'
-- Including context 'longdistance' in context
'local'
-- Including context 'default' in context 'local'
-- Including context 'parkedcalls' in context
'local'
-- Including context 'trunklocal' in context
'local'
-- Including context 'trunkld' in context 'local'
-- Including context 'iaxtel700' in context
'local'
-- Including context 'trunktollfree' in context
'local'
-- Including context 'iaxprovider' in context
'local'
-- Including context 'room' in context 'local'
-- Including context 'demo' in context 'default'
[pbx_wilcalu.so] => (Wil Cal U (Auto Dialer))
Entered Wil-Calu fd=37
[pbx_spool.so] => (Outgoing Spool Support)
/var/spool/asterisk/outgoing
[app_dial.so] => (Dialing Application)
== Registered application 'Dial'
[app_playback.so] => (Trivial Playback Application)
== Registered application 'Playback'
[app_voicemail.so] => (Comedian Mail (Voicemail
System))
== Registered application 'VoiceMail'
== Registered application 'VoiceMailMain'
[app_directory.so] => (Extension Directory)
== Registered application 'Directory'
[skipping app_intercom.so]
[app_mp3.so] => (Silly MP3 Application)
== Registered application 'MP3Player'
[app_system.so] => (Generic System() application)
== Registered application 'System'
[app_echo.so] => (Simple Echo Application)
== Registered application 'Echo'
[app_record.so] => (Trivial Record Application)
== Registered application 'Record'
[app_image.so] => (Image Transmission Application)
== Registered application 'SendImage'
[app_url.so] => (Send URL Applications)
== Registered application 'SendURL'
[app_disa.so] => (DISA (Direct Inward System Access)
Application)
== Registered application 'DISA'
[app_agi.so] => (Asterisk Gateway Interface (AGI))
== Registered application 'EAGI'
== Registered application 'AGI'
[app_qcall.so] => (Call from Queue)
[app_adsiprog.so] => (Asterisk ADSI Programming
Application)
== Registered application 'ADSIProg'
[app_getcpeid.so] => (Get ADSI CPE ID)
== Registered application 'GetCPEID'
[app_milliwatt.so] => (Digital Milliwatt (mu-law)
Test Application)
== Registered application 'Milliwatt'
[app_zapateller.so] => (Block Telemarketers with
Special Information Tone)
== Registered application 'Zapateller'
[app_datetime.so] => (Date and Time)
== Registered application 'DateTime'
[app_setcallerid.so] => (Set CallerID Application)
== Registered application 'SetCallerID'
[app_festival.so] => (Simple Festival Interface)
== Registered application 'Festival'
[app_queue.so] => (True Call Queueing)
== Registered application 'Queue'
== Manager registered action Queues
== Manager registered action QueueStatus
== Registered application 'AddQueueMember'
== Registered application 'RemoveQueueMember'
[app_senddtmf.so] => (Send DTMF digits Application)
== Registered application 'SendDTMF'
[app_parkandannounce.so] => (Call Parking and
Announce Application)
== Registered application 'ParkAndAnnounce'
[app_striplsd.so] => (Strip trailing digits)
== Registered application 'StripLSD'
[app_setcidname.so] => (Set CallerID Name)
== Registered application 'SetCIDName'
[app_lookupcidname.so] => (Look up CallerID Name from
local database)
== Registered application 'LookupCIDName'
[app_substring.so] => (Save substring digits in a
given variable)
== Registered application 'SubString'
[app_macro.so] => (Extension Macros)
== Registered application 'Macro'
[app_authenticate.so] => (Authentication Application)
== Registered application 'Authenticate'
[app_softhangup.so] => (Hangs up the requested
channel)
== Registered application 'SoftHangup'
[app_lookupblacklist.so] => (Look up Caller*ID
name/number from blacklist
database)
== Registered application 'LookupBlacklist'
[app_waitforring.so] => (Waits until first ring after
time)
== Registered application 'WaitForRing'
[app_privacy.so] => (Require phone number to be
entered, if no CallerID
sent)
== Registered application 'PrivacyManager'
[app_db.so] => (Database access functions for
Asterisk extension logic)
== Registered application 'DBget'
== Registered application 'DBput'
== Registered application 'DBdel'
== Registered application 'DBdeltree'
[app_chanisavail.so] => (Check if channel is
available)
== Registered application 'ChanIsAvail'
[app_enumlookup.so] => (ENUM Lookup)
== Registered application 'EnumLookup'
[app_voicemail2.so] => (Comedian Mail (Voicemail
System))
== Registered application 'VoiceMail2'
== Registered application 'VoiceMailMain2'
[app_transfer.so] => (Transfer)
== Registered application 'Transfer'
[app_zapras.so] => (Zap RAS Application)
== Registered application 'ZapRAS'
[app_meetme.so] => (Simple MeetMe conference bridge)
== Registered application 'MeetMeCount'
== Registered application 'MeetMe'
[app_flash.so] => (Flash zap trunk application)
== Registered application 'Flash'
[app_zapbarge.so] => (Barge in on Zap channel
application)
== Registered application 'ZapBarge'
[codec_g723_1.so] => (Annex A (fixed point)
G.723.1/PCM16 Codec Translator)
== Registered translator 'g723tolin' from format 0
to 6, cost 128
== Registered translator 'lintog723' from format 6
to 0, cost 861
[codec_g723_1b.so] => (Annex B (floating point)
G.723.1/PCM16 Codec
Translator)
== Registered translator 'g723tolinb' from format 0
to 6, cost 51
== Registered translator 'lintog723b' from format 6
to 0, cost 235
[codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec
Translator)
== Registered translator 'ilbctolin' from format 10
to 6, cost 8
== Registered translator 'lintoilbc' from format 6
to 10, cost 46
[codec_gsm.so] => (GSM/PCM16 (signed linear) Codec
Translator)
== Registered translator 'gsmtolin' from format 1 to
6, cost 2
== Registered translator 'lintogsm' from format 6 to
1, cost 5
[codec_mp3_d.so] => (MP3/PCM16 (signed linear)
Translator (Decoder only))
== Registered translator 'mp3tolin' from format 4 to
6, cost 17
[codec_lpc10.so] => (LPC10 2.4kbps (signed linear)
Voice Coder)
== Registered translator 'lpc10tolin' from format 7
to 6, cost 6
== Registered translator 'lintolpc10' from format 6
to 7, cost 8
[codec_adpcm.so] => (Adaptive Differential PCM
Coder/Decoder)
== Registered translator 'adpcmtolin' from format 5
to 6, cost 1
== Registered translator 'lintoadpcm' from format 6
to 5, cost 1
[codec_ulaw.so] => (Mu-law Coder/Decoder)
== Registered translator 'ulawtolin' from format 2
to 6, cost 1
== Registered translator 'lintoulaw' from format 6
to 2, cost 1
[codec_alaw.so] => (A-law Coder/Decoder)
== Registered translator 'alawtolin' from format 3
to 6, cost 1
== Registered translator 'lintoalaw' from format 6
to 3, cost 1
[codec_a_mu.so] => (A-law and Mulaw direct
Coder/Decoder)
== Registered translator 'alawtoulaw' from format 3
to 2, cost 1
== Registered translator 'ulawtoalaw' from format 2
to 3, cost 1
[format_gsm.so] => (Raw GSM data)
== Registered file format gsm, extension(s) gsm
[format_wav.so] => (Microsoft WAV format (8000hz
Signed Linear))
== Registered file format wav, extension(s) wav
[format_mp3.so] => (MPEG-1,2 Layer 3 File Format
Support)
== Registered file format mp3, extension(s)
mp3|mpeg3
[format_wav_gsm.so] => (Microsoft WAV format
(Proprietary GSM))
== Registered file format wav49, extension(s) WAV
[format_vox.so] => (Dialogic VOX (ADPCM) File Format)
== Registered file format vox, extension(s) vox
[format_pcm.so] => (Raw uLaw 8khz Audio support
(PCM))
== Registered file format pcm, extension(s)
pcm|ulaw|ul|mu
[format_g729.so] => (Raw G729 data)
== Registered file format g729, extension(s) g729
[format_pcm_alaw.so] => (Raw aLaw 8khz PCM Audio
support)
== Registered file format alaw, extension(s) alaw|al
[format_h263.so] => (Raw h263 data)
== Registered file format h263, extension(s) h263
[format_jpeg.so] => (JPEG (Joint Picture Experts
Group) Image Format)
== Registered format 'jpg' (JPEG (Joint Picture
Experts Group))
[cdr_csv.so] => (Comma Separated Values CDR Backend)
[chan_oh323.so] => (OpenH323 Channel Driver)
0:00.006 OpenH323 Wrapper OpenH323
Wrapper Version
0.0alpha0 by inAccess Networks
(www.inaccessnetworks.com) on Unix Linux
(2.4.20-8-i686) at 2003/8/8
3:36:01.570
== OpenH323 Channel Ready (v0.5.4)
Asterisk Ready.
New Thread 1215102256 (LWP 560)]
[New Thread 1223494960 (LWP 561)]
[New Thread 1231887664 (LWP 562)]
[New Thread 1240280368 (LWP 563)]
[New Thread 1248673072 (LWP 564)]
== OpenH323 Channel Ready (v0.5.4)
Asterisk Ready.
*CLI> [New Thread 1257065776 (LWP 565)]
[New Thread 1265458480 (LWP 566)]
-- Called g3/19258467700
-- Zap/1-1 answered H323:25514
[New Thread 1273851184 (LWP 567)]
[New Thread 1282243888 (LWP 568)]
[New Thread 1290636592 (LWP 569)]
[New Thread 1299029296 (LWP 570)]
[New Thread 1307422000 (LWP 571)]
-- Playing 'pbx-transfer'
-- Unable to find extension '1 ' in context
'local'
-- Playing 'pbx-invalid'
Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread 1273851184 (LWP 567)]
0x42074d60 in _int_realloc () from /lib/tls/libc.so.6
BackTrace of core file
[root@node-40244e3d root]# gdb -cv core.14144
gdb: unrecognized option `-cv'
Use `gdb --help' for a complete list of options.
[root@node-40244e3d root]# gdb -c core.14144
GNU gdb Red Hat Linux (5.3post-0.20021129.18rh)
Copyright 2003 Free Software Foundation, Inc.
GDB is free software, covered by the GNU General
Public License, and you are
welcome to change it and/or distribute copies of it
under certain
conditions.
Type "show copying" to see the conditions.
There is absolutely no warranty for GDB. Type "show
warranty" for details.
This GDB was configured as "i386-redhat-linux-gnu".
Core was generated by `/usr/sbin/asterisk -vvvcdg'.
Program terminated with signal 11, Segmentation fault.
#1 0x42074d60 in ?? ()
Configuration of OpenH323 channel driver
----------------------------------------
Version: 0.5.4
Listening on address: 0.0.0.0:9090
Gatekeeper used: OpenGate@node-40244e3.net
FastStart/H245Tunnelling/H245inSetup: ON/ON/ON
Supported format(s): G.723.1 GSM G.711U G.711A G.729A
Jitter buffer limits (min/max): 50-200
TCP port range: 5000 - 31000
UDP (RAS) port range: 5000 - 31000
UDP (RTP) port range: 10000 - 20000
IP Type-of-Service value: 0
User input mode: 2
Max number of inbound H.323 calls: 100
Max number of outbound H.323 calls: 100
extension.conf
---------------------------------------
Context 'default' created by 'pbx_config' ]
'11' => 1. Dial(OH323/011)
[pbx_config]
'12' => 1. Dial(OH323/012)
[pbx_config]
2.CHANGES IN asterisk -vvc when a call from PBX goes
through using 711 codec
---------------------------------------------------------------------------
) on Unix Linux (2.4.20-8-i686) at 2003/8/8
5:12:11.416
== Registered channel type 'OH323' (OpenH323 Channel
Driver)
== OpenH323 Channel Ready (v0.5.4)
== Parsing '/etc/asterisk/enum.conf': Found
Asterisk Ready.
*CLI> NOTICE[1256017200]: File chan_zap.c, Line 4135
(ss_thread): Got event
2 (Ring/Answered)...
WARNING[1264610608]: File chan_oh323.c, Line 2154
(alerted_h323_connection):
Call with reference 8019 in unexpected
state (4).
0:43.967 H225 Caller:81229a8 H225
Received connect PDU.
*** [ip$localhost/8019] H.323 CONTROL PROTOCOL ERROR
(Roundtrip Delay)
oh323 show info
Information about active OpenH323 channel(s)
--------------------------------------------
Num. Token State Init
RX/TX Format
Remote RTP Addr. Local RTP Addr.
0 ip$localhost/8019 ESTABLI Local
320/240 G.711U
61.11.XX.XXX:16386 6XX.XX.XX.XX:10000
*CLI> 2:46.707 H225 Caller:81229a8 H225
Read error (0):
== Spawn extension (default, 12, 1) exited non-zero
on 'Zap/1-1'
2:49.747 H323 Cleaner H323
Connection ip$localhost/8019
terminated.
3.)Changes in the asterisk -vvc when call fails to go
through PBX after
diabling 711 codec
----------------------------------------------------------------------------
-----------
== Parsing '/etc/asterisk/enum.conf': Found
Asterisk Ready.
*CLI> NOTICE[1256017200]: File chan_zap.c, Line 4135
(ss_thread): Got event
2 (Ring/Answered)...
0:47.213 H225 Caller:8128950 H245 Write
PDU fail: no control
channel.
0:47.235 H323 Cleaner H323
Connection
ip$localhost/15853 terminated.
ERROR[1256017200]: File chan_oh323.c, Line 711
(oh323_call): H323:0: Could
not call 012.
WARNING[1256017200]: File codec_gsm.c, Line 165
(gsmtolin_framein): Invalid
GSM data
WARNING[1256017200]: File codec_gsm.c, Line 165
(gsmtolin_framein): Invalid
GSM data
WARNING[1256017200]: File codec_gsm.c, Line 165
(gsmtolin_framein): Invalid
GSM data
...
....
....
....
Information about active OpenH323 channel(s)
--------------------------------------------
<No active H.323 connections>
*CLI> oh323 show stats
Statistics of OpenH323 channel driver
-------------------------------------
Up since: Fri Aug 8 05:20:00 2003
Inbound H.323 calls: 0
Outbound H.323 calls: 0
Dropped inbound H.323 calls: 0
Blocked outbound H.323 calls: 0
Total inbound H.323 calls detected: 0
Total outbound H.323 calls attempted: 1
H.323 call errors: 0
H.323 answer errors: 0
----------------------------------------------
----- Original Message -----
From: "Michael Manousos"
<manousos@inaccessnetworks.com>
To: <asterisk-users@lists.digium.com>
Sent: Friday, August 08, 2003 3:56 PM
Subject: Re: [Asterisk-Users] Problem
-ATA-711-723-Oh323-Asterisk
>
> Sip Rtp wrote:
> > Hi List,
> >
> > I am facing the reverse problem as stated here.I
am> > using ATA 186 to make
> > and recieve call to * through OH323 driver.
> > When I use G711 codec in the ATA to make call then
> > then as soon as i dial an
> > extension the * crashes with 'segmentation fault'.
>
> More information is needed.
> You should provide a backtrace of the core file,
> the screen log of Asterisk (generated when executed
> with "asterisk -vvvcdg"), your oh323.conf and the
important> sections of extensions.conf.
>
> > But the same scenerio works fine when i use 723
codec> > in the ATA .I can dial
> > the number and extension very well/(I have 723
support> > in the * ).
> > But now problem comes in the outbound as when i
use a> > extension like
> > exten=>12,1,Dial(OH323/12)
> > Then the call goes through but i don't hear any
voice.> > So my two problems are
> > 1.Why asterisk gives seg. fault when i dial exten
on> > 711 codec from ATA
> > 2.Why can't i hear voice from * to ATA when i use
723> > in ATA.
> > for 2nd i think that there is mismatch between the
> > codecs so can we change
> > the priority order of the codecs used in the * or
> > Oh323 and if yes, then
> > how?
> >
> > Please ask if any further Input is required.
> >
> > Rgds
> > Manoj K Gupta
> >
>
>
> Michael.
>
>
>
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>
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